Issues 28000 - 28999

[..]
ASTERISK-28000: sample: PJSIP endpoint identifier order doesn't match reality
ASTERISK-28001: res_pjsip_registrar: Improve performance of inbound handling
ASTERISK-28002: When T.140 realtime text is negociated, a lot of debug traces are generated
ASTERISK-28003: Qualifying non-authenticated endpoints on startup
ASTERISK-28004: dns: Core ast_dns_get_nameservers does not support configured IPv6 servers
ASTERISK-28005: channel.c: ARI ring only once
ASTERISK-28006: PJSIP: Missing "party=calling"/"party=called" in Remote-Party-ID
ASTERISK-28007: rtcp-mux is put in SDP answer regardless of offer
ASTERISK-28008: Asterisk crash signal 11, Segmentation fault
ASTERISK-28009: Queue predial for callee channel
ASTERISK-28010: PJSIP: Crash with MWI implicit subscription replaced by explicit
ASTERISK-28011: chan_sip: get_refer_info() attempted unlock mutex 'peer' without owning it!
ASTERISK-28012: ODBC Voicemail and 'pollmailboxes=yes' does not update shared state via XMPP
ASTERISK-28013: res_http_websocket: Crash when reading HTTP Upgrade requests
ASTERISK-28014: Can't record video with Record application
ASTERISK-28015: pjsip: bad file descriptor when passing pjsip qualify endpoint to standard out
ASTERISK-28016: PJSIP sends duplicate 183 Progress responses
ASTERISK-28017: I'm using Chan-Sip but PJSIP errors/warnings.
ASTERISK-28018: IP Fragmentation happening instead of DTLS fragmentation on handshake server hello certificate
ASTERISK-28019: Crash in ast_format_get_sample_rate when play an audio
ASTERISK-28020: res_pjsip_transport_websocket: Properly set 'received' for IPv6
ASTERISK-28021: backtrace.c: New crash due to double-free.
ASTERISK-28022: res_pjsip realtime: uri column in ps_contacts table can be too short
ASTERISK-28023: CONFBRIDGE maybe break when playing annoucement
ASTERISK-28024: my queue members (eg. SIP/{SIPuser name}/{phone no}) showing invalid. Is it neccessary to registered all the agents of queue in sip.conf. Or please give me alternative .Please help me with the issue
ASTERISK-28025: Asterisk webrtc in SIPML5
ASTERISK-28026: Session timers not updating after reload for active calls
ASTERISK-28027: Call Setup Crash ILBC -> ULAW
ASTERISK-28028: Disk I/O error, dropped calls
ASTERISK-28029: [patch] res_musiconhold : music on hold will not start if previous hold just reached end of file
ASTERISK-28030: pbx_lua: Deadlock when reloading module
ASTERISK-28031: Invalid UTF-8 string - Element Block
ASTERISK-28032: Realtime queuemembers are not updated during retry phase
ASTERISK-28033: AMI event "NewExten" is set to the wrong class
ASTERISK-28034: chan_sip unstable with TLS after asterisk start or reloads
ASTERISK-28035: PJSIP Error
ASTERISK-28036: Codec negotiation when incoming re-INVITE has no SDP
ASTERISK-28037: How to set runing REST API after hungup a call
ASTERISK-28038: Queue log is incorrect in Attended transfer &Blind transfer
ASTERISK-28039: Null pointer crash for ast_stream_get_type
ASTERISK-28040: pbx: "dialplan reload" is removing minus symbol from dynamic hints
ASTERISK-28041: Asterisk freezes unexpected
ASTERISK-28042: Asterisk freezes unexpected
ASTERISK-28043: Error 4 in app_queue.so
ASTERISK-28044: res_stasis : random crash related to ast_channel_varset_type
ASTERISK-28045: configure script does not enforce libunbound2 version
ASTERISK-28046: Remove stale nonoptreq references
ASTERISK-28047: chan_pjsip: Declined video stream is added when no video codecs configured and session refresh with removed video stream occurs
ASTERISK-28048: res_pjsip fails to migrate endpoint devstate from Unavailable to Not in use after restart until pjsip reload (or reregister)
ASTERISK-28049: res_pjproject build failure
ASTERISK-28050: Asterisk crash with Abort error
ASTERISK-28051: RTP engine should only accept audio frames with allowed payloads
ASTERISK-28052: app_voicemail: Voicemail help plays conflicting options
ASTERISK-28053: chan_pjsip: Wrong or missing Q.850 reason in CANCEL
ASTERISK-28054: Asterisk Core dumping on regular basis 13.21.1
ASTERISK-28055: app_queue: Per-member wrapup time missing from AddQueueMember application
ASTERISK-28056: res_pjsip: Incorrect endpoint status after endpoint synchronization for a specific AOR
ASTERISK-28057: chan_sip: SipNotify via AMI behaves differently to CLI
ASTERISK-28058: Set busy on soft phone
ASTERISK-28059: PJSIP: Update bundled PJPROJECT to version 2.8
ASTERISK-28060: Queue answered out of order
ASTERISK-28061: Creating new realtime pjsip endpoint not updating state
ASTERISK-28063: unable to register IP extension in FreePBX
ASTERISK-28064: Memory leak issue in json.c, endpoints.c and stasis_channels.c While using AMI
ASTERISK-28065: res_odbc: missing SQL error diagnostic
ASTERISK-28066: modification of modules.conf
ASTERISK-28067: res_pjsip_sdp_rtp: Extra fingerprint attribute in SDP
ASTERISK-28068: Wrong button label in german dpma localization
ASTERISK-28069: Dropping CDRs records with local languages
ASTERISK-28070: testsuite: Sniffer assumes pjmedia will use ports below 10000
ASTERISK-28071: chan_sip: ignores the "fromdomain" option if "fromuser" option is presented
ASTERISK-28072: app_agent_pool: Crash when heavily manipulated externally using AMI
ASTERISK-28073: asterisk memory leak
ASTERISK-28074: Run Two Exec command in asterisk application
ASTERISK-28075: Null pointer in SRTP Handling Crash
ASTERISK-28076: bridging: Asterisk crashes when receiving an empty realtime text frame
ASTERISK-28077: res_pjsip: improve realtime performance on CLI 'pjsip show contacts'
ASTERISK-28078: pjsip: Missing support for TLS CRL
ASTERISK-28079: [patch]core: New variables CONNECTED_LINE_ORIGINAL_* for interception routine CONNECTED_LINE_SEND_SUB
ASTERISK-28080: AsteriskNOW
ASTERISK-28081: chan_sip: Asterisk 12+ chan_sip doesn't report AST_CEL_PICKUP in handle_invite_replaces
ASTERISK-28082: res_ari_bridges: allow the app to be specified on bridge creation when using ARI
ASTERISK-28083: res_agi: Asterisk truncates result of get_variable to 1024 characters
ASTERISK-28084: app_queue: QueueMemberStatus Event flooding AMI
ASTERISK-28085: testsuite: Figure out why chan_sip blind transfer tests are failing.
ASTERISK-28086: chan_pjsip: Crash when initiating PlayDTMF over AMI
ASTERISK-28087: add flag to allow CALLERID(num) to be placed in Contact header in chan_pjsip
ASTERISK-28088: ast_restart: Test is failing occasionally
ASTERISK-28089: function ast_sendtext() create RTP realtime packets with a trailing null byte in the payload
ASTERISK-28090: chan_sip: MessageSend() doesn't use configured SIP peers
ASTERISK-28091: Control endpoints media_address use per call basis
ASTERISK-28092: res_pjsip. AMI event Registry. Field Cause is empty
ASTERISK-28093: pbx: Deadlock from holding channel lock when it shouldn't be
ASTERISK-28094: pjsip. Disable anonymous for local sip domains and force to inbound registration
ASTERISK-28095: func_odbc: Crash when calling an ODBC function from another ODBC function
ASTERISK-28096: pjsip: PJPROJECT 2.8 causes test failure
ASTERISK-28097: Queue option 'b' + SIP_HEADER make issue
ASTERISK-28098: ATTENDED_TRANSFER_COMPLETE_SOUND deadlocks in Local channels (Asterisk 11)
ASTERISK-28099: When I try to find the state of Endpoints it is showing unavailable
ASTERISK-28100: how to disable native_dahdi techonology for bridge
ASTERISK-28101: Unable to load config file 'statsd.conf'
ASTERISK-28102: stasis: Use implementation specific cache for channel snapshots
ASTERISK-28103: stasis: Filter messages at publishing to reduce work done
ASTERISK-28104: AstriCon Feedback: Automatically create a 1 line dialplan context for stasis apps
ASTERISK-28105: AstriCon Feedback: Allow multiple connections for the same Stasis application
ASTERISK-28106: Astricon Feedback: Unable to filter ARI events when GETting causes overload of events
ASTERISK-28107: app_confbridge: Participant info labels aren't being added to the SDPs
ASTERISK-28108: Deadlock in publish_msg (stasis.c)
ASTERISK-28109: pbx_dundi: Does not support chan_pjsip
ASTERISK-28110: rtp: Incorrect Packetization
ASTERISK-28111: build: CHANGES/UPGRADE are irritating to work with.
ASTERISK-28112: Asterisk is not able to use newly released mysql connector odbc 8.0.12 for voicemail
ASTERISK-28113: gerrit: Minor Tweak to email template to support dark theme.
ASTERISK-28114: Random crash
ASTERISK-28115: rtp: Cache channel snapshot locally
ASTERISK-28116: stasis: Investigate automatic disabling of message creation
ASTERISK-28117: stasis: Add statistics for usage when in developer mode
ASTERISK-28118: Not forwarding RTP packets / Packet loss
ASTERISK-28119: stasis: Segment channel snapshot to reduce creation cost
ASTERISK-28120: stasis: Audit loitering topics
ASTERISK-28121: Don't play the early media when I have an incoming call
ASTERISK-28122: app_queue.so crashed
ASTERISK-28123: stun keeps revaluating
ASTERISK-28124: Is Asterisk SAP Gateway supported in Pakistan, Are you offering a Cloud solution for SAP Gateway
ASTERISK-28125: app_queue: Revert broken queue channel reference patch
ASTERISK-28126: Can't login into asterisk.org
ASTERISK-28127: Buffer overflow for DNS SRV/NAPTR records
ASTERISK-28128: postgresql config db upgrade fail at alembic upgrade fe6592859b85
ASTERISK-28129: Incorrect Behavior for rewrite_contact when Re-Invite omits routset
ASTERISK-28130: when calling from the queue, all contacts are not called
ASTERISK-28131: ${ANSWEREDTIME} is incorrect in Dial app in Asterisk 15.6
ASTERISK-28132: res_pjsip_registrar: Asterisk crashing with large number of PJSIP registration
ASTERISK-28133: Realtime iaxfriends table 'port' definition causing issues
ASTERISK-28134: Legacy forums using Symantec certs throw warning on Chrome
ASTERISK-28135: Opus Codec Parked Calls Drop
ASTERISK-28136: Allow the sip_to_pjsip script to be used in a pipe
ASTERISK-28137: res_pjsip_notify: improve realtime performance on CLI completion on the endpoint
ASTERISK-28138: sip_to_pjsip.py does not convert setvar to set_var
ASTERISK-28139: RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls
ASTERISK-28140: repeated segmentation faults
ASTERISK-28141: mysql same value calldate,answer,end
ASTERISK-28142: res_agi: asterisk will execute continuation of agi file in situation of simultaneous hangup in both side (caller and callee)
ASTERISK-28143: app_amd: Infinite loop on silent calls
ASTERISK-28144: [patch] New function PJSIP_PARSE_URI to parse an URI and return a specified part of the URI
ASTERISK-28145: SRTp with bria and zoiper
ASTERISK-28146: pbx_config: Only the first [globals] section is processed.
ASTERISK-28147: Unable to connect to Asterisk from asterisk-java
ASTERISK-28148: MoH restart at each Dial
ASTERISK-28149: PJSIP: Setting CallerID for outbound channel from predial handler doesn't work
ASTERISK-28150: Formatting error in documentation
ASTERISK-28151: app_voicemail: MWI fails with mailboxes=##@device instead of mailboxes=##@default
ASTERISK-28152: mysql two record
ASTERISK-28153: [patch] chan_sip: fix Reason-Phrase for 603 Response
ASTERISK-28154: stasis: Add support for shutting down topic
ASTERISK-28155: res_pjsip_mwi: Crash at shutdown due to order problem
ASTERISK-28156: Race condition involving session->media (res_pjsip_session) leads to crash.
ASTERISK-28157: Asterisk crashes when the res_pjsip_* modules unload
ASTERISK-28158: Some conditions prevent running of el_end, break the terminal.
ASTERISK-28159: SIGABRT caused by stack corruption in hashkeys_read when no matching keys present
ASTERISK-28160: Asterisk was failed down and restarted again.
ASTERISK-28161: Removal of Previous Patch Causes PJSIP Timer Issues
ASTERISK-28162: [patch] need to reset DTMF last sequence number and timestamp on RTP renegotiation
ASTERISK-28163: Asterisk with 2 nic's
ASTERISK-28164: stasis: Improve channel snapshot segmenting
ASTERISK-28165: app_queue: QueueMemberStatus ami event duplication
ASTERISK-28166: app_voicemail: Asterisk unresponsive after changing voicemail password with ODBC
ASTERISK-28167: 256 cipher during outgoing calls
ASTERISK-28168: app_queue: Adding a blank entry into sql queue_members crashes asterisk.
ASTERISK-28169: ARI /channels/create handler causes core dump
ASTERISK-28170: ARI POST /channels/{channelId}/dial memory leak
ASTERISK-28171: res_resolver_unbound: DNS issue when under load, can't make outgoing calls
ASTERISK-28173: Deadlock in chan_sip handling subscribe request during res_parking reload
ASTERISK-28174: PJSIP issue with TEL (RFC 3966)
ASTERISK-28175: PJSIP support for TEL (RFC 3966)
ASTERISK-28176: Make usage tracking switchable
ASTERISK-28177: block calling from contextA to contextB
ASTERISK-28178: Program terminated with signal 11, Segmentation fault.
ASTERISK-28179: Asterisk responses 100 trying about 4 seconds after sending INVITE.
ASTERISK-28180: ami: High memory (increased upto 15 GBs) while pushing 500 calls continuously for 7 hours
ASTERISK-28181: ari: Originating overwrites channel start time
ASTERISK-28182: chan_pjsip: When connected_line_method is set to invite, asterisk is not trying UPDATE
ASTERISK-28183: Channel SIP message notifier
ASTERISK-28184: Asterisk hangs up receiving unexpected frame format
ASTERISK-28185: chan_pjsip: Subsequent same responses are not stopped
ASTERISK-28186: stasis: Filter messages at publishing based on to_* presence
ASTERISK-28187: Asterisk 16.0 crash on receiving fax
ASTERISK-28188: mysql db import error
ASTERISK-28189: configured pjsip endpoints go offline when a new endpoint registers
ASTERISK-28190: Asterisk Crashing
ASTERISK-28191: features.conf option blindxfer doesn't seem to do Pattern Matching.
ASTERISK-28192: Use kamailio and asterisk on call about 500 current
ASTERISK-28193: auto gain control
ASTERISK-28194: chan_sip: Leak using contact ACL
ASTERISK-28195: stasis: Don't create subscription change messages if noone cares
ASTERISK-28196: bridge_softmix: Does not support WebRTC source with multi video tracks.
ASTERISK-28197: stasis: ast_endpoint struct holds the channel_ids of channels past destruction in certain cases
ASTERISK-28198: res_ari: Add new hangup causes for ARI Channel DELETE command
ASTERISK-28199: sdp: iLBC codec does not contain "mode=" attribute
ASTERISK-28200: res_rtp_asterisk: Duplicate DTMF with endpoint when media received in between
ASTERISK-28201: [patch] confbridge: no announce to the marked users when they join an empty conference
ASTERISK-28202: pjproject: fails to build on ppc64el
ASTERISK-28203: Asterisk 13.13 crash/restart bug
ASTERISK-28204: Asterisk Restarted | Crashes chan_sip.c
ASTERISK-28205: module app_queue.so stoped
ASTERISK-28206: Read Application is NOT correct in single channel
ASTERISK-28207: promiscredir
ASTERISK-28208: chan_pjsip: 183 without SDP followed by 180 does not result in media
ASTERISK-28209: res_pjsip_t38: Passthrough causes crash when re-invite collision
ASTERISK-28210: res_pjsip_outbound_registration: Registrations reports "Rejected" for No Response type failures
ASTERISK-28211: chan_pjsip: Path header is not used with PJSIP_DIAL_CONTACTS
ASTERISK-28212: stasis: Statistics broke ABI under developer mode
ASTERISK-28213: res_pjsip: Threads pile up needlessly when AOR is blocked
ASTERISK-28214: PJSIP and CHAN_SIP issues
ASTERISK-28215: app_voicemail: Leaving voicemail sometimes doesn't trigger NOTIFYs
ASTERISK-28216: Crash when race condition between manager_play_dtmf and ast_hangup
ASTERISK-28217: mwi broken?
ASTERISK-28218: app_queue: Asterisk crashes when using Queue with a pre-dial handler (option b)
ASTERISK-28219: res_ari: Channel create and dial may cause "BUG! Must supply a channel name.." error
ASTERISK-28220: SPY extension follow the channel after being bridged
ASTERISK-28221: Bug in ast_coredumper
ASTERISK-28222: Regression: MWI polling no longer works
ASTERISK-28223: Configure failure
ASTERISK-28224: res_parking: ParkAndAnnounce hangs up call when lot is full.
ASTERISK-28225: app_voicemail: Channel variable VM_MESSAGEFILE not updated correctly if message marked "urgent"
ASTERISK-28226: I want to know about how to connect mysql to store CDR
ASTERISK-28227: Adding more ARI subscription type
ASTERISK-28228: res_pjsip: pjsip show contacts prints double entries
ASTERISK-28229: Asterisk not responding/reloading
ASTERISK-28230: res_rtp_asterisk: abs-send-time extension added with Asterisk 15.5.0 breaks GXV3140 video telephony
ASTERISK-28231: res_http_websocket: Not responding to Connection Close Frame (opcode 8)
ASTERISK-28232: core: RAII using clang use-after-scope issue
ASTERISK-28233: pbx_dundi: PJSIP is not a supported technology
ASTERISK-28234: pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
ASTERISK-28235: ERROR[-1]: app_voicemail.c:2836 inboxcount2: Couldn't find mailbox in context
ASTERISK-28236: Support separated HTTP request
ASTERISK-28237: "FRACK!, Failed assertion bad magic number" happens when unsubscribe an application from an event source
ASTERISK-28238: PJSIP realtime. getcontext not working with DUNDI
ASTERISK-28239: Device_state - Change of returned status
ASTERISK-28240: Unexpected unhold when Asterisk cannot find the moh-files
ASTERISK-28241: Call pickup fails, if dialed from subroutine, but succeeds with macro
ASTERISK-28242: Can't Retrieve Voicemail from PostgreSQL database when "msgnum" is INT type
ASTERISK-28243: Corrupted SIP after handling a 302 redirect
ASTERISK-28244: stasis: Filter messages at publishing to AMI/ARI
ASTERISK-28245: DMTF emulation not working if direct_media = yes and dtmf_mode = info is set
ASTERISK-28246: Support skipping on the g726 format
ASTERISK-28247: res_ari: Applications not being cleaned up after certain scenarios
ASTERISK-28248: testsuite: Write tests for ARI 'move'
ASTERISK-28249: res_monitor: Segfault with Monitor(wav,file,i)
ASTERISK-28250: build: Cross-compilation fails for target arm-linux-gnueabihf
ASTERISK-28251: CI: Fix CI so it reverifies commit message changes
ASTERISK-28252: HangupHandler manager events are never thrown
ASTERISK-28253: res_pjsip_session: Adding rtcp stats result into the session
ASTERISK-28254: testsuite: PJSIP tests can't tolerate retransmissions (and they happen sometimes)
ASTERISK-28255: res_rtp_asterisk: REMB RTCP packet sending may be incorrect
ASTERISK-28256: Video plays back in slow motion
ASTERISK-28257: res_http_websocket: PING / PONG opcodes break data reception
ASTERISK-28258: DUNDi Does Not Register chan_pjsip Realtime Endpoints On Register
ASTERISK-28259: CLONE - app_voicemail: Leaving voicemail sometimes doesn't trigger NOTIFYs
ASTERISK-28260: Asterisk segfault when rtp negotiation is wrong or fails
ASTERISK-28261: PJSIP: SDP in 180 Ringing is ignored
ASTERISK-28262: Custom device state losing - (dash)
ASTERISK-28263: codec_opus: errors setting max_playback_rate and bitrate to "sdp"
ASTERISK-28264: Added topic_all container
ASTERISK-28265: PJSIP show channelstats incorrect information output
ASTERISK-28266: Added ARI resource /ari/aserisk/ping
ASTERISK-28267: res_stasis: Add ability to switch applications
ASTERISK-28268: Asterisk + create new app + conncet mysql
ASTERISK-28269: chan_mobile can't connect to phone again
ASTERISK-28270: Creating call using ARI, does not increasing/decreasing the call count(core show calls)
ASTERISK-28271: Opensuse Leap 15 --with-jannson-bundled will not compile
ASTERISK-28272: The basic-pbx config samples don't produce a running asterisk
ASTERISK-28273: H245 logical channels don't close when asterisk is terminated the call.
ASTERISK-28274: Asterisk 11 crashes randomly
ASTERISK-28275: Error saving agent on queue_log on pickup calls with * 8
ASTERISK-28276: TESTTIME feature not working
ASTERISK-28277: database: Add some basic logging
ASTERISK-28278: Asterisk extensions periodically lost registrations
ASTERISK-28279: Added creation timestamp for bridge
ASTERISK-28280: chan_sip problem with registration when challenge contains a "domain" field with protocol.
ASTERISK-28281: Enabling cli command bridge destroy <bridge id>
ASTERISK-28282: AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in chan_sip)
ASTERISK-28283: Fixed wrong description for RTCPReceived/RTCPSent
ASTERISK-28284: switching between native_bridge and simple_bridge can cause one way audio
ASTERISK-28285: Fixed wrong RTT calculation
ASTERISK-28286: chan_sip - no lock pvt data in proc_session_timer()
ASTERISK-28287: Pjsip rewrite port in from/to headers on reply
ASTERISK-28288: Resources (udptl fd) leaking for T.38 calls
ASTERISK-28289: Feature: Allow detection of inband progress for outbound channels
ASTERISK-28290: res_resolver_unbound.so: Failed to perform async DNS resolution
ASTERISK-28291: res_pjsip_path: Not applied on OPTIONS requests
ASTERISK-28292: Changed to show all channel stats including wrong media
ASTERISK-28293: New Build ast_expr2fz.o Will Not Link
ASTERISK-28294: Segmentation Fault on strchr
ASTERISK-28295: chan_sip / pjsip: Non UTF-8 handling could be better
ASTERISK-28296: Getting Error While Running asterisk -rvvv
ASTERISK-28297: cdr_engine - taskprocessor.c
ASTERISK-28298: chan_iax2: Does not update endpoint state in all cases
ASTERISK-28299: 481 Call/Transaction Does Not Exist on receiving MESSAGE event
ASTERISK-28300: AST_PBX_MAX_STACK is too low for some applications
ASTERISK-28301: Allow voicemail boxes to be subscribed to with a presence event package
ASTERISK-28302: ARI: "Error destroying mutex" when listing all ARI applications
ASTERISK-28303: res_rtp_asterisk: Interaction between smoother and DTMF can cause out of order timestamps
ASTERISK-28304: app_voicemail. Issue with NOTIFYs
ASTERISK-28305: register_aor_core: Unable to bind contact
ASTERISK-28306: res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be sent
ASTERISK-28307: Segmentation fault in libasteriskpj.so.2
ASTERISK-28308: Double line at pjsip show contacts
ASTERISK-28309: res_pjsip: Wrong Contact and Via fields with multiple UDP interfaces
ASTERISK-28310: Inconsistent Stasis/AMI event ordering for queue members calling
ASTERISK-28311: dsp: ast_dsp_silence_noise_with_energy wrong judgment of frame format
ASTERISK-28312: res_pjsip_diversion: Corrupted SIP Diversion field after handling a 302 redirect
ASTERISK-28313: ARI: userevents should be delivered via AMI too
ASTERISK-28314: ARI: API changed but "apiVersion" in rest-api\resources.json did not
ASTERISK-28315: Asterisk is crash in Amazon EC2, i recompile with --disable BUILD_NATIVE, but is crash anyway
ASTERISK-28316: Asterisk Restarting / Creating core dump file
ASTERISK-28317: Add logical group at DAHDIChannel event and create "dahdi_group" at CHANNEL function
ASTERISK-28318: Conference Bridge Options (via AMI) Setting Bug
ASTERISK-28319: musl: Crash on startup when loading modules
ASTERISK-28320: Added ARI resource /ari/channels/{channelid}/rtp_statistics
ASTERISK-28321: res_rtp_asterisk: Fixing possible divide by zero for rtcp stat calculation
ASTERISK-28322: chan_pjsip: Add option to allow ignoring of 183 without SDP
ASTERISK-28323: pjsip: sip.conf to pjsip.conf conversion script fails
ASTERISK-28324: res_mwi_devstate does not allow variable replacement in dialplan
ASTERISK-28325: Command to delete a bridge
ASTERISK-28326: ari: Added timestamp for some ari events.
ASTERISK-28327: res_pjsip_endpoint_identifier_ip: Missing IdentifyDetail event
ASTERISK-28328: MeetMe global non-admin mute is muting admins that subsequently join
ASTERISK-28329: RTCP - Error building JSON
ASTERISK-28330: call gets disconnected every 30 sec after answer
ASTERISK-28331: Add an option in MeetMe to start the conference with all non-admins muted
ASTERISK-28332: Variable ALTCONF ignored when service is used in Debian
ASTERISK-28333: StasisEnd event makes wrong timestamp value
ASTERISK-28334: pjsip : direct_media options doesnt work
ASTERISK-28335: stasis: Make topic and maybe subscription names unique and more useful
ASTERISK-28336: After ARI continue, hangup() application does not create SoftHangupRequest event
ASTERISK-28337: Regular segmentation faults
ASTERISK-28338: Asterisk crashes with ERROR *** /usr/sbin/asterisk': corrupted size vs. prev_size: 0x00007f77400cfcf0 *** when there are 48 SIP outbound calls.(almost 144 SIP Channels)
ASTERISK-28339: Added ARI resource /ari/channels/{channelid}/dump
ASTERISK-28340: MWI NOTIFY is not sent immediately after a VM was created/deleted
ASTERISK-28341: res_config_odbc eliminates empty custom (“@” prefix) variables
ASTERISK-28342: Ast-to-Ast setup using the same rtcpinterval crashes RTCP and audio stream
ASTERISK-28343: Added app_name, app_data to channel type
ASTERISK-28344: Wrong music on hold handling on multi party attendant transfer
ASTERISK-28345: IMS TEL URI incoming INVITE RFC 3966 not recognized
ASTERISK-28346: useless transcoding
ASTERISK-28347: ari: Crash while deleting bridge with channels in it
ASTERISK-28348: Failed to initialize OOH323 endpoint-OOH323 Disabled
ASTERISK-28349: Pause reason not reported in QueueMember AMI event
ASTERISK-28350: manager: Stasis backed up due to locking
ASTERISK-28351: CDR disposition incorrectly logged at file when AMD classifies as MACHINE
ASTERISK-28352: German language sounds referenced by app-voicemail missing
ASTERISK-28353: stasis: Crash at shutdown when statistics enabled
ASTERISK-28354: app_queue: Call to Unavailable member when ringinuse=yes and another member is available
ASTERISK-28355: Trunk to Inbound works for a few seconds only
ASTERISK-28356: app_queue: CLI set ringinuse for realtime member not working
ASTERISK-28357: Fixing duplicated subscription adding
ASTERISK-28358: Too much allocations in frame.c
ASTERISK-28359: User login page does not apply any form of price determination
ASTERISK-28360: User login page does not apply any form of price determination
ASTERISK-28361: app_confbridge: Missing MIXMONITOR_FILENAME in ConfBridge AMI
ASTERISK-28362: strtok_r() makes gcc compile warning
ASTERISK-28363: Millisecond-resolution call stats including PDD in channel variables
ASTERISK-28364: func_strings: HASHKEYS in shared variable space cannot be retrieved
ASTERISK-28365: New ARI for application execution.
ASTERISK-28366: Add timeout for response to StasisStart-event
ASTERISK-28367: Servname not supported for ai_socktype / Could not resolve socket address
ASTERISK-28368: Low performance. Many errors taskprocessor
ASTERISK-28369: app_queue: Member device state "invalid" when second call is ringing and hint is used
ASTERISK-28370: res_pjsip_t38: Not accepting Audio Re-invite after T.38 rejection
ASTERISK-28371: chan_pjsip: DTMF Mode auto_info fallback lead to both inband and info
ASTERISK-28372: Asterisk REPLY Wrong Contact header port (TCP)
ASTERISK-28373: persistant registrations not working for TCP transport
ASTERISK-28374: latest asterisk unconditionally launch gcc --version, even if the compiler is different
ASTERISK-28375: res_pjsip: New configuration setting to allow disabling norefersub
ASTERISK-28376: After changing parameter "media_address" in sip.conf to any valid IP, we are getting one way or no voice issues on few calls. Sometimes voice passes successfully, we would like to know why this is happening and resolution for same.
ASTERISK-28377: ARI: Crash when unanswered channel rejoins dial bridge automatically
ASTERISK-28378: Added detail subscriber/subscription info for stasis show app cli
ASTERISK-28379: pjsip: show channelstats incorrect information output
ASTERISK-28380: Asterisk CLI Showing ERRORS: frame.c: Excessive refcount 100000 reached on ao2 object when total calls reached almost 30000 count.
ASTERISK-28381: MWI indicators not updating correctly on version 15.7.2
ASTERISK-28382: app_confbridge: Leave message not played when penultimate person leaves
ASTERISK-28383: Contact status is reported as REACHABLE when contact is deleted
ASTERISK-28384: res_xmpp: Crash when distribute_events=yes and res_mwi_devstate loads
ASTERISK-28385: res_ari_channels: Added detail hangup code settings
ASTERISK-28386: res_pjsip does not follow DNS SRV priority values
ASTERISK-28387: res_pjsip: Contact status latency is not pushed through the AMI
ASTERISK-28388: Endpoint sync causes device unreach when a new contact is added
ASTERISK-28389: PJ ICE Rx error status code: 370400 'Bad Request'.
ASTERISK-28390: Duplicate contacts appear when running "asterisk -rx 'pjsip list contacts'"
ASTERISK-28391: res_indications: Crash requesting autocomplete on indications cli command
ASTERISK-28392: The no-partial-inlining flag isn't passed to the bundled pjproject or jansson builds
ASTERISK-28393: Multidomain support issue
ASTERISK-28394: sip_outbound_publish_client_add_publisher Failed assertion bad magic number
ASTERISK-28395: Asterisk occasionally fails to hangup channels
ASTERISK-28396: missing voicemail option in sql table for realtime
ASTERISK-28397: ARI Push Configuration - duplicate entries
ASTERISK-28398: Column order of contrib realtime MySQL config for sippeers causes issues with NAT
ASTERISK-28399: channel.c: Exceptionally long queue length queuing
ASTERISK-28400: res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for transport-cc
ASTERISK-28401: app_confbridge: Add *_all remb behavior variants
ASTERISK-28402: res_pjsip_registrar: SEGV in registrar_find_contact
ASTERISK-28403: Add native Prometheus support to Asterisk
ASTERISK-28404: astobj2.c: FRACK, memory leakage
ASTERISK-28405: ChanSpy : cannot hangup spying channel if no audio is received
ASTERISK-28406: PJSIP compose invalid request-URI in ACK for re-INVITE
ASTERISK-28407: Segfault in hash_ao2_find_next at astobj2_hash.c:581
ASTERISK-28408: Asterisk crashes intermittently if use sipML5 as the SIP client, regardless of the total number of peers in use.
ASTERISK-28409: unexpected build problem at install procedure
ASTERISK-28410: Crash on unload res_pjsip_mwi.so and show subscriptions
ASTERISK-28411: Build link error bundled pjproject : relocation against symbol cant be used when shared object
ASTERISK-28412: GCC 9 catches more string formatting issues
ASTERISK-28413: pjsip show channelstats crashes asterisk while printing a channel being hung up
ASTERISK-28414: Asterisk crash on internal calls
ASTERISK-28415: segfault: sprint_list_entry (entry=entry@entry=0x7f9e30b4d8b0, line=line@entry=0x7f9e70676590 "\340ggp\236\177", len=256) at res_pjsip_history.c:669
ASTERISK-28416: Unable to get rtp codec payload code for slin
ASTERISK-28417: SDP negotiation issue
ASTERISK-28418: Timezone Problem
ASTERISK-28419: app_amd: Does not work with silence suppression
ASTERISK-28420: In WebRTC video call scenario, packet loss lead to frozen video。
ASTERISK-28421: Wrong type used for timestamp in res_rtp_asterisk
ASTERISK-28422: Memory Leak in Confbridge menu
ASTERISK-28423: ARI causes STASIS Deadlock
ASTERISK-28424: Task processor queue regularly fills up with subscriptions, using curl config and external mwi, test phone doesn't stay registered
ASTERISK-28425: Realtime Voicemail locks Asterisk when no filesystem folder exists
ASTERISK-28426: Address out of bounds in ast_str_hash
ASTERISK-28427: new mwi.h include missing from some dahdi source files, causes build failure
ASTERISK-28428: app_dial: Incorrectly alters Hangup and DialEnd events when the c argument is passed, but app_dial didn't cancel the call.
ASTERISK-28429: Bad answer JSON when request ARI
ASTERISK-28430: res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
ASTERISK-28431: Asterisk memory increases and CentOS OEM killer kills Asterisk process
ASTERISK-28432: ODBC
ASTERISK-28433: More than 1 AMI connection ends up dying
ASTERISK-28434: Segfault: INTERNAL_OBJ (user_data=0xffffffff) at astobj2.c:131
ASTERISK-28435: cdr_pgsql: Unix socket doesn't work
ASTERISK-28436: Transcoding happening even though it is not necessary
ASTERISK-28437: Taskprocessor doesn't process the tasks
ASTERISK-28438: chan_pjsip: segfault in pj_grp_lock_acquire
ASTERISK-28439: When we dial from a number the phone routes to a phone on the system instead of the number we dial
ASTERISK-28440: pjsip: configure does not detect LibreSSL
ASTERISK-28441: fax: T38 fallback to voice does not change codec
ASTERISK-28442: stasis_state: Create a stasis module to cache last known state
ASTERISK-28443: app_voicemail: remove dependency on stasis cache
ASTERISK-28444: chan_pjsip: Peer IP for SSL handshake errors not logged
ASTERISK-28445: res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on hangup when TEST_FRAMEWORK enabled
ASTERISK-28446: Asterisk NOTIFY NOT_INUSE and UNAVAILABLE both send same XML dialog <state>terminated</state>
ASTERISK-28447: res_pjsip_messaging: In-dialog MESSAGE with no body causes crash
ASTERISK-28448: chan_sip: Sometimes G729 (without annexb=no) is negotiated
ASTERISK-28449: scheduler: Supported time can be exceeded by PUBLISH
ASTERISK-28450: Program terminated with signal 11, Segmentation fault at t38_gateway.c:2189
ASTERISK-28451: chan_pjsip: HANGUPCAUSE(<channel>,tech) fails to get SIP cause for rejected calls
ASTERISK-28452: pjsip: <sess-version> of SDP is not incremented though SDP may be changed on reinvite without SDP offer
ASTERISK-28453: Voicemail: Failed to Lock Path: File Exists
ASTERISK-28454: res_fax.c UTF-8 validation for remotestationid and pbx_builtin_setvar_helper
ASTERISK-28455: res_odbc: Connection through proxysql fails
ASTERISK-28456: Asterisk crashed and core dumps when attempting to free a frame
ASTERISK-28457: [patch] Fix crash in chan_dahdi on 32-bit systems caused by ASTERISK-28317
ASTERISK-28458: res_pjsip_sdp_rtp: Remove unused variable
ASTERISK-28459: pbx: Asterisk is dropping part of a string passed to functions
ASTERISK-28460: res_pjsip_sdp_rtp: Fix ICE candidate leak with specific usage
ASTERISK-28461: Crash in app.Pickup
ASTERISK-28462: func_talkdetect: TALK_DETECT firing immediately even if phone microphone is muted
ASTERISK-28463: res_pjsip_path: Crash when invalid contact is configured
ASTERISK-28464: FRACK!, Failed assertion with res_pjsip
ASTERISK-28465: Broken SDP can cause a segfault in a T.38 reINVITE
ASTERISK-28466: Typo on the IVR System webpage
ASTERISK-28467: 404 Link on Asterisk Wiki Dashboard
ASTERISK-28468: 404 on What is Asterisk webpage
ASTERISK-28469: AMI frozen on high load
ASTERISK-28470: Mutex deadlock in audio_audiohook_write_list
ASTERISK-28471: I have dial IVR number using asterisk AMI how can i send DTMF automatically
ASTERISK-28472: Asterisk occasionally passes a NULL as srtp->session to srtp_protect/unprotect causing SEGV
ASTERISK-28473: res_pjsip_t38: Crash on Asterisk 16.4
ASTERISK-28474: Old Book link on Asterisk Books webpage
ASTERISK-28475: Segmentation fault
ASTERISK-28476: Asterisk Deadlock During chan_pjsip Call Transfer
ASTERISK-28477: Crash when not specifying "dbfile" in res_config_sqlite3.conf
ASTERISK-28478: Crash performing "core reload" with modified res_config_sqlite3.conf
ASTERISK-28479: Crash when no database specified using driver "sqlite3" in extconfig.conf
ASTERISK-28480: json integer overflow in ssrc and timestamp
ASTERISK-28481: Push notification using Asterisk server
ASTERISK-28482: Asterisk 13.22.0 Segmentation fault PJSIP TLS+SRTP about 60 endpoints
ASTERISK-28483: packet lost on UDPTL wrap around
ASTERISK-28484: Add AudioSocket support
ASTERISK-28485: Program terminated with signal 11, Segmentation fault.
ASTERISK-28486: Randomly generated segfault in asterisk process
ASTERISK-28487: compile menuselect on gentoo
ASTERISK-28488: pjsip mwi: n+1 sip notify's sent on re-register
ASTERISK-28489: Channel variable SIPFROMDOMAIN for chan_pjsip to setup From header URI domain
ASTERISK-28490: Segfault in chan_pjsip in grp_lock_dec_ref
ASTERISK-28491: Allow in and out file descriptors to be used in AGI - Create XAGI
ASTERISK-28492: pjsip reload not reloading wizard endpoint/pickup_group endpoint/call_group
ASTERISK-28493: Can't log into community.asterisk.org
ASTERISK-28494: Facing Segmentation fault (core dumped) identified by "asterisk -cvvvvvvv". I am also unable to connect remote asterisk because of this issue. Please provide solution. Thank you
ASTERISK-28495: res_pjsip_t38: 200 OK with SDP answer with declined stream causes crash
ASTERISK-28496: PJSIP max_retries
ASTERISK-28497: func_odbc: truncating Unicode string on readsql
ASTERISK-28498: cel / cdr: Event times may be incorrect
ASTERISK-28499: translate: Crash when frame does not have a "src" field set
ASTERISK-28500: CDR Endtime is coming lesserthan the CDR Starttime
ASTERISK-28501: Can't log into community.asterisk.org
ASTERISK-28502: chan_pjsip incorrectly re-writes REGISTER 200 Response Contact
ASTERISK-28503: Asterisk sudden crashes with segmentation fault
ASTERISK-28504: Asterisk is crashing too frequently whenever a large number of PJSIP AOR are trying to register on asterisk.
ASTERISK-28505: app_voicemail/IMAP: segfault in leave_voicemail because not checking mailstream
ASTERISK-28506: asterisk crashes at random frequent intervals
ASTERISK-28507: Wiki docs missing for MessageWaiting
ASTERISK-28509: PJSIP cnonce generated on Linux contains 36 characters, NEC only supports up to 32 characters
ASTERISK-28510: Asterisk crashing on dtls handshake
ASTERISK-28511: codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32
ASTERISK-28512: Add pass-through support for H.265 (HEVC) codec
ASTERISK-28513: Should To: be rewritten when forwarding to a phone
ASTERISK-28514: phoneprov and RealTime
ASTERISK-28515: res_pjsip: TLS close notify alert is not sent before closing the connection
ASTERISK-28516: Crash Under AMI Taskprocessor Backup
ASTERISK-28517: Asterisk segfault in t38_interpret_parameters at res_pjsip_t38.c:457
ASTERISK-28518: chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up Dahdi Call On Hold
ASTERISK-28519: [issue] Asterisk
ASTERISK-28520: Failed to create temporary storage
ASTERISK-28521: pjsip: Memory Leak
ASTERISK-28522: chan_pjsip does not support fallback from t.38 to fax over alaw/ulaw
ASTERISK-28523: Asterisk 16.5.0 Memory leak
ASTERISK-28525: chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up
ASTERISK-28526: Error executing SQL (COMMIT): database is locked
ASTERISK-28527: ChanIsAvail() creates a CDR if unanswered=yes is set in cdr.conf
ASTERISK-28528: Local channel video stream broken with ConfBridge and first_marked=yes
ASTERISK-28529: Segfault in res_pjsip_pubsub.c due to accessing a destroyed dialog
ASTERISK-28530: Help in making Simultaneous calls NOT Bridged
ASTERISK-28531: fetch all incoming calls mobile number to our extenstions
ASTERISK-28532: Segfault at res_rtp_multicast.c:211 (function set_type)
ASTERISK-28533: func_jitterbuffer: Add support for video synchronization
ASTERISK-28534: Segmentation fault when there is no priority for an extension
ASTERISK-28535: Error when change my callerid(num)
ASTERISK-28536: Asterisk release candidates fail to build on FreeBSD
ASTERISK-28537: Different music on hold queue, 1 for ringing and 1 for putting customer on hold
ASTERISK-28538: chan_pjsip: Deadlock on fax detection
ASTERISK-28539: Failed t.38 negotiation when B leg sends t.38 re-invite
ASTERISK-28540: Deadlock In Stasis/ARI
ASTERISK-28541: Asterisk 16.5.0 Memory leak
ASTERISK-28542: [patch] add the ability for asterisk to generate on-hold re-invites
ASTERISK-28543: When Asterisk cannot connect to SIP socket it starts to flood with "Bad descriptor" errors and hangs
ASTERISK-28544: Wrong contact representation in ipv6 mode
ASTERISK-28545: Introduce a 'playlist' mode for res_musiconhold
ASTERISK-28546: MWI Leaks
ASTERISK-28547: PJSIP AOR configuration "contact=sip:" crash
ASTERISK-28548: PJSIP received a 183 with sendonly will be onhold
ASTERISK-28549: Two repeated 183
ASTERISK-28550: SDP maxptime
ASTERISK-28551: IPv4 address in SDP o= is (null) when configured for NAT using pjsip
ASTERISK-28552: res_pjsip_mwi: Frack during unload on unsolicited_mwi container
ASTERISK-28553: stasis.c: Crash during unload
ASTERISK-28554: [patch] Add recipes for sample Queues
ASTERISK-28555: Unable to Register WebRTC client when using a Proxy
ASTERISK-28556: pjsip blind transfer fails
ASTERISK-28557: dtmf detection
ASTERISK-28558: BridgeAttendedTransfer not received if audio is playing in a bridge
ASTERISK-28559: Making Asterisk work with Amtelco Genesis Software
ASTERISK-28560: do_monitor Thread hangs on 99% cpu and doesn't respond
ASTERISK-28561: Asterisk Deadlocks
ASTERISK-28562: SIP WSS message not processed until next frame arrives
ASTERISK-28563: Additional configuration [extName](+) not always working
ASTERISK-28564: Memory leak with pjsip 2.9 and SIPS / SRTP
ASTERISK-28565: Conference is disconnecting after entering conference PIN number
ASTERISK-28566: CDR backend unload problem during active call(s)
ASTERISK-28567: Problem with ASTERISK-20207: Asterisk should clear out any .lock files in the voice mail directory on startup.
ASTERISK-28568: Potential memory leak on core reload
ASTERISK-28569: Missing check for variable buf in function config_text_file_load in utils/extconf.c
ASTERISK-28570: Potential infinite loop in function find_matching_priority
ASTERISK-28571: cdr_pgsql: accesses obsolete (and finally removed) column
ASTERISK-28572: Memory leaks in res_calendar_exchange and res_calendar_icalendar
ASTERISK-28573: Missing event AgentComplete on AttendedTransfer
ASTERISK-28574: pjproject fails to build on 16.6.0, works on 16.5
ASTERISK-28575: MWI Send Notify Crash on 16.6
ASTERISK-28576: res_rtp_asterisk: ICE Completion Crash when sent packet length doesn't match
ASTERISK-28578: race condition on pjsip channelstats command
ASTERISK-28580: Bypass SYSTEM write permission in manager action allows system commands execution
ASTERISK-28581: How to integrate gcloud speech recognition with asterisk ivr in python
ASTERISK-28582: Breaking out of a long playing video when using Background
ASTERISK-28583: web gui loop
ASTERISK-28584: Configure direct_media=yes in pjsip. Conf ,don't valid, Media still flows asterisk
ASTERISK-28585: ari/resource_events: Crash in event session cleanup
ASTERISK-28586: Typo in README-SERIOUSLY.bestpractices.md
ASTERISK-28587: Asterisk crash when answering a call
ASTERISK-28588: MODULE PROVIDING BELOW SECTION(S): res_pjsip_acl
ASTERISK-28589: chan_sip: Depending on configuration an INVITE can alter Addr of a peer
ASTERISK-28590: utils.c throws repeated warnings; "pthread_attr_setstacksize: Invalid argument"
ASTERISK-28591: We have two server with debian os and install asterisk on that, we need server redundancy, how it possible.
ASTERISK-28592: Asterisk Crash (Segmentation fault)
ASTERISK-28593: Fix check on forwarded voicemail-to-email message body
ASTERISK-28594: Chan_SIP Crash (Segmentation fault)
ASTERISK-28595: Asterisk 15.7.2 with TLS
ASTERISK-28596: 34/5000 Problem with the sip in asterisk 16
ASTERISK-28597: every 2 to 5 mints my asterisk will be stop
ASTERISK-28598: Configure Fax for receiving in asterisk 13.20
ASTERISK-28599: Problem with the sip in asterisk 16
ASTERISK-28600: Unable to configure webrtc
ASTERISK-28601: bridge: BRIDGEPVTCALLID and BRIDGEPEER emtpy after Dial(SIP/...)
ASTERISK-28602: res_pjsip_outbound_registration: Maximum retries reached
ASTERISK-28603: Presence subscription on Cisco SIP phone needs special Cisco-styled XML - PJSIP
ASTERISK-28604: app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
ASTERISK-28605: chan_dahdi: Deadlock in Hangup Scenarios with concurrent command pri show span X
ASTERISK-28606: Caller Id unknown Showing to receiver
ASTERISK-28607: P-Asserted-Identity value ${EXTEN}
ASTERISK-28608: app_amd: Use time calculation to calculate timeout
ASTERISK-28609: Memory Leak in res_rtp_asterisk.c
ASTERISK-28610: CDR fields in second leg use wrong variables from first leg
ASTERISK-28611: sip.conf host!=dynamic peer specific options (e.g. directmedia=off, transport=tcp) not working!?
ASTERISK-28612: res_pjsip_t38: crash on reinvite with zero port and no c= line
ASTERISK-28613: func_curl: CURLOPT cannot set Content-Type header
ASTERISK-28614: app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead of only "sending"
ASTERISK-28615: chan_dahdi: PRI span status may stay "Down, Active" after a short alarm
ASTERISK-28616: parking: Deadlock when multi call parking
ASTERISK-28617: DTMF over SIP INFO in scenarios without audio does not work well
ASTERISK-28618: bridge_softmix: hold not cleared when joining a softmix bridge
ASTERISK-28619: data mismatching in "queue show" CLI
ASTERISK-28620: Segfault in chan_pjsip on pj_strcmp when filtering a transmit message
ASTERISK-28621: Enforce T.38 error correction mode at 200 ok received
ASTERISK-28622: Differences in gcc options cause the undefined sanitizer to fail in pjproject when using dev-mode
ASTERISK-28623: pjsip: PJPROJECT_CONFIGURE_OPTS install location not honored
ASTERISK-28624: res_pjsip_outbound_registration: add SRV failover
ASTERISK-28625: Playback of local files impacted by large media cache
ASTERISK-28626: Missing arguments in PJSIP_CONTACT function documentation
ASTERISK-28627: Error FRACK!, failed assertion bad magic number
ASTERISK-28628: Debian 10.2: Warning when app_voicemail is compiling
ASTERISK-28629: [patch] Add an "inhibitCOLP" flag to the bridges REST API
ASTERISK-28630: Asterisk crash
ASTERISK-28631: res_parking: Doesn't park when parkee and parker are the same
ASTERISK-28632: ConfBridge spawns record_command before MixMonitor ends
ASTERISK-28633: stasis bridge topic leak
ASTERISK-28634: Invite loop within PJSIP
ASTERISK-28635: res_rtp_asterisk should allow for acl style whitelist/blacklist of ICE candidates
ASTERISK-28636: app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR.
ASTERISK-28637: chan_sip+native_bridge_rtp: directmedia compatibility check failure when negociated ptime is not default ptime.
ASTERISK-28638: Simplify dialplan for Dial, Page, and ChanIsAvail
ASTERISK-28639: res_pjsip_endpoint_identifier_ip: Add ability to match on source port
ASTERISK-28640: app_voicemail with ODBC - error logging is useless
ASTERISK-28641: res_pjsip Segfaults when realtime configuration to an AOR points to a not existent AOR
ASTERISK-28642: SayAlpha not working in GoSub
ASTERISK-28643: Deadlock, possibly in Parking, maybe in combination with AMI status messages.
ASTERISK-28644: Stale comment in app_queue about ring_entry exception
ASTERISK-28645: Menuselect Asterisk
ASTERISK-28646: asterisk segfault at sp error 4 in libsrtp.so.0.0.0
ASTERISK-28647: chan_sip: RTP frames not transmitted after emitting a COLP
ASTERISK-28648: chan_sip/chan_pjsip copy_via_headers() function not RFC 3261 compliant
ASTERISK-28649: Segfault: ast_variables_destroy #channel_vars #set_var #sorcery_realtime
ASTERISK-28650: Voicemail Build Options change not documented
ASTERISK-28651: chan_sip logs errors on tx to non-existent TCP connections
ASTERISK-28652: unable to call outbound cli call
ASTERISK-28653: call through test call file not happening
ASTERISK-28654: Enabling Real Time Text (RTT) for PJSIP library
ASTERISK-28655: core: Many things require an audio stream to be present to work
ASTERISK-28656: improve pjproject.conf sample configuration
ASTERISK-28657: SIPS TLS connection fails when session ticket extension is used
ASTERISK-28658: app_confbridge: Add support for setting maximum sample rate
ASTERISK-28659: res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them
ASTERISK-28660: res_fax: wrap Asterisk initiated negotiation with config option
ASTERISK-28661: chan_iax jitterbuffer growing when time sources not in sync
ASTERISK-28662: Asterisk install does not compile res_pjsip/config_transport.c
ASTERISK-28663: jansson: Support old versions
ASTERISK-28664: "trustrpid" is misspelled in sip_to_pjsip.py
ASTERISK-28665: No clientcertificate requested
ASTERISK-28666: Integration with FXO/FXS GAteway via FXO line/OPenvox card
ASTERISK-28667: Asterisk ignores parsing of config files if a Byte order mark is present
ASTERISK-28668: aymmetric_rtp_codec dialplan function
ASTERISK-28669: chan_sip: Device states lost when sip reload
ASTERISK-28670: astspooldir setting in asterisk.conf isn't configurable
ASTERISK-28671: Outbound Channel to endpoint that negotiates iLBC mode = 20 produces Warning in CLI and one way audio
ASTERISK-28672: the sky is falling
ASTERISK-28673: GET FULL VARIABLE documentation clarification
ASTERISK-28674: Asterisk becomes unstable after SS7 signalling link restarts
ASTERISK-28675: Ho we could record both leg of single call separately in wav file format.
ASTERISK-28676: How we could store DTMF in variable
ASTERISK-28677: CDR billsec is always 0 for transferred calls
ASTERISK-28679: stasis application is destroyed after its creation
ASTERISK-28680: Incorrect results from ast_sorcery_changeset_create()
ASTERISK-28681: Agent can login queue, but this queue member status is "Invalid"
ASTERISK-28682: app_record: Lack of `beep` audio file causes application to return error and hangup
ASTERISK-28683: No clientcertificate requested
ASTERISK-28684: SIPAddHeader ignored when sending sip MESSAGE
ASTERISK-28685: check_expr2: linking (when hardening) and cross-compiling troubles
ASTERISK-28686: chan_sip strictrtp=yes fails when media source is changed: no audio
ASTERISK-28687: Originate with early media option treats the call as answered which does not reflect the actual state of the call
ASTERISK-28688: Matching SIP TCP peer by IP with insecure=port regression
ASTERISK-28689: res_pjsip: Crash when locking group lock when sending stateful response
ASTERISK-28690: I just want to test something, please ignore
ASTERISK-28691: unknown codec resulting a call dropped
ASTERISK-28692: cdr: Asterisk crashed after NoOp application in realtime
ASTERISK-28693: chan_sip: SIP MESSAGE beginning with a whitespace appears empty in the dialplan
ASTERISK-28694: res_phoneprov: Phone provision stopped updating with a reload of res_phoneprov
ASTERISK-28695: core: minmemfree watermark uses free RAM, not available RAM
ASTERISK-28696: PJSIP exception when parsing 'Via' header
ASTERISK-28697: res_pjsip: Named ACL does not update on reload if changed
ASTERISK-28698: func_cdr: Getting CDR variables always returns success with a string
ASTERISK-28699: ast_coredumper does not find asterisk running process and silently fails
ASTERISK-28700: Page system with music for swimming pool
ASTERISK-28701: app_queue: Core reload resets queue stats, even when keepstats=yes
ASTERISK-28702: chan_dahdi: holding a channel via flash to dialtone times out after 0:16:40
ASTERISK-28703: PRI-STATUS DOWN
ASTERISK-28704: AMI QueuePause cannot find interface
ASTERISK-28705: chan_sip: Phones loose abiltiy to work, core restarting asterisk fixes issue
ASTERISK-28706: silk 24hHz doesn't show up in 'core show translation' output
ASTERISK-28707: Pjsip Threadpool cant handle more than 10 calls per second
ASTERISK-28708: app_queue: Deadlock with "queue show" and "shared_lastcall" option
ASTERISK-28709: pjproject: Bundled pjproject install error
ASTERISK-28710: Should be able to disable the /httpstatus URI in the built-in HTTP server
ASTERISK-28711: HangupCause shows misleading information on timeout
ASTERISK-28712: Possible freeze in app_queue
ASTERISK-28713: res_stasis_playback: Error building JSON
ASTERISK-28714: REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults
ASTERISK-28715: Setting Content-Type header in MessageSend application by using PJSIP
ASTERISK-28716: ICE: pjnath shouldn't wait for ICE to complete before allowing sending
ASTERISK-28717: ARI Data Model DialplanCEP - Missing required properties
ASTERISK-28718: chan_sip: Returns 403 if RTP ports are depleted, should return 503
ASTERISK-28719: Cannot remove defaultrule from queue using realtime queues
ASTERISK-28720: When using realtime queues penaltymemberslimit checks for all members count
ASTERISK-28721: Asterisk restarts with core dump
ASTERISK-28722: I agree
ASTERISK-28723: How do I WebRTC PeerConnection with two Asterisk WebRTC end points
ASTERISK-28724: AWS Ubuntu : PJSIP call drops after 30 seconds
ASTERISK-28725: Bridge error on incoming calls on asterisk 16.8.0 and 13.31.0
ASTERISK-28726: install_prereq script uses the interactive mode when installing aptitude
ASTERISK-28727: Some of stasis messages don't contain asterisk_id
ASTERISK-28728: Asterisk crash in RTP stack (segfault)
ASTERISK-28729: i can't make a call in my goautodial 4 server
ASTERISK-28730: res_pjsip_session: Fix out of order session refreshes
ASTERISK-28731: Directmedia Reinvites have SDP with codecs from configuration not negotiation
ASTERISK-28733: stream: Add support for adding/removing streams during SFU/calls
ASTERISK-28734: Segmentation fault when calling ast_format_get_codec_id
ASTERISK-28735: Realtime MoH Unknown format '' -- defaulting to SLIN
ASTERISK-28736: Asterisk periodic restarts when executing sip reload.
ASTERISK-28737: Asterisk 13.28.0 repeatable crashes
ASTERISK-28738: Incorrect state machine used when MOH_PASSTHRU is used
ASTERISK-28739: Dropping redundant connected line update
ASTERISK-28740: Issue regarding Sip account registeration
ASTERISK-28741: Issue regarding Sip account registeration
ASTERISK-28742: res_rtp_asterisk: static for audio due to incomplete dtls/srtp setup
ASTERISK-28743: Asterisk is crashing if the 200 OK with SDP
ASTERISK-28744: No transfer events logged in queue_log
ASTERISK-28745: [BOUNTY] support_path missing after reload
ASTERISK-28746: res_pjsip_outbound_registration keeps retrying the first entry in a SRV record set
ASTERISK-28747: YES/NO attributes are not set properly when creating PJSIP sorcery objects via ARI
ASTERISK-28748: Recording failed when making many calls per second
ASTERISK-28749: Matching on Caller ID not working if the dialled extension is a pattern
ASTERISK-28750: TLS/SSL Key too small error
ASTERISK-28751: Difficulty sending 16k slin16 back to Asterisk via External Media
ASTERISK-28752: Support receiving both t38 and t30 faxes
ASTERISK-28754: ASTERISK-28738 Causes Audio Issue After Hold
ASTERISK-28755: SIP/Stasis: SIP headers not transmitted in the "variables" field
ASTERISK-28756: Codec Negotiation: add incoming_call_offer_pref option
ASTERISK-28757: 403 - Forbidden auth ID when SIP messaging Anveo
ASTERISK-28758: pjsip startup errors when using "with-ssl" configure option
ASTERISK-28759: A non negotiated rtp frame causes call disconnection when there is a SSRC change
ASTERISK-28760: G729a codec can't be loaded
ASTERISK-28761: Assigning CallerIDNum from DNIDDigits to chan_pjsip
ASTERISK-28762: Problem setting up ssl connection. Internal SSL error
ASTERISK-28763: Issue with SQLBindParameter with ODBC on StrLen_or_IndPtr
ASTERISK-28764: res_rtp_asterisk: Improve NACK support and seqno handling
ASTERISK-28765: tcptls API: bad file descriptor when connection fails
ASTERISK-28766: PJSIP blind transfer not completed after using Proceeding()
ASTERISK-28767: chan_pjsip: Caller ID not used when checking for extension, callerid supplement executed too late
ASTERISK-28768: No audio on remote NATted phone when using local_net behind another NATted Asterisk
ASTERISK-28769: DTLS Handshake Fails to Occur if ice_support is enabled but not used
ASTERISK-28770: res_pjsip: AVC denial with default SELinux setup on CentOS 7
ASTERISK-28771: Unable to install asterisk in Ubuntu 12/14/16/18.04 versions
ASTERISK-28772: Add indication tones for Indonesia
ASTERISK-28773: Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
ASTERISK-28774: chan_pjsip's rtptimeout is erroneously triggered during direct-media (native_rtp) bridge
ASTERISK-28775: SRV fix for ASTERISK-28746 fails with Deutsche Telecom
ASTERISK-28776: Non async-signal-safe syscalls used after fork before exec
ASTERISK-28777: Codec Negotiation: add outgoing_call_offer_prefs option
ASTERISK-28778: Public IP in contact URI when softphone traffic goes through VPN
ASTERISK-28779: what will be the process to send dtmf after call bridge using agi.
ASTERISK-28780: app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup
ASTERISK-28781: res_config_odbc: Sorcery doesn't set default value if the ODBC realtime value is blank
ASTERISK-28782: Add support for Content-Disposition header in multi-part INVITES
ASTERISK-28783: res_pjsip_session: Allow default non-audio streams to have reflected state
ASTERISK-28784: res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
ASTERISK-28785: chan_local: Unnecessary transcoding for Originate (local channels)
ASTERISK-28786: Duplicated Endpoints, AORS, Auths user ARI sorcery
ASTERISK-28787: res_pjsip_session: Decide more intelligently when to add video
ASTERISK-28788: func_aes: incorrectly printing error 'declined to load'
ASTERISK-28789: test_utils: incorrectly printing error 'declined to load'
ASTERISK-28790: Crash during conference call using confbridge and video
ASTERISK-28791: Manager Action MixMonitorMute not working
ASTERISK-28792: codec_gsm: while building, optimization flag is overwritten
ASTERISK-28793: Asterisk 13.32.0 crash in pjsip_tx_data_add_ref
ASTERISK-28794: res_pjsip: Crash when escaping during URI printing
ASTERISK-28795: channel: write to a stream on multi-frame writes
ASTERISK-28796: func_channel: cannot read fields exten, context, userfield, channame from dialplan
ASTERISK-28797: [patch] tcptls: Fix notice when TLS is enabled but not configured.
ASTERISK-28798: [patch] chan_sip: TCP/TLS client without server.
ASTERISK-28799: I have issue
ASTERISK-28800: core: SIGSEGV on DTMF when some modules not loaded
ASTERISK-28801: [patch] stasis: Avoid always true warnings with clang.
ASTERISK-28802: ari: /dial function disconnect Inbound call automatically if Outbound is "busy" or "discard" in case delay "Answer" message.
ASTERISK-28803: [patch] chan_unistim: Avoid tautological warnings with clang.
ASTERISK-28804: [patch] app_osplookup.c: Avoid a format truncation.
ASTERISK-28805: follow me with empty database fileds broken after res odbc fix for empty strings
ASTERISK-28807: ICES dialplan issue
ASTERISK-28808: [patch] test_stasis: Avoid always true warning with clang.
ASTERISK-28809: [patch] res_rtp_asterisk: Avoid absolute value on unsigned subtraction.
ASTERISK-28810: Segmentation fault in ast_manager_build_channel_state_string_prefix
ASTERISK-28811: Crash occurs when fax session switches from T.38 to audio
ASTERISK-28812: First DTMF is not get
ASTERISK-28813: func_volume: Allow decimal numbers as parameter to improve granularity
ASTERISK-28814: Asterisk stops processing SIP requests because of an undetermined reason
ASTERISK-28815: res_pjsip.so segfaulting on 17.3.0
ASTERISK-28816: [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
ASTERISK-28817: chan_pjsip: constant DTMF tone if RTP is not setup yet
ASTERISK-28818: [patch] BuildSystem: Allow space in path.
ASTERISK-28819: [patch] bridge_softmix_binaural: Show state in menuselect.
ASTERISK-28820: pjproject_bundled: Modes Developer+Noisy give Stop.
ASTERISK-28821: a code change to chan_pjsip breaks SIP/ISUP internetworking in early state
ASTERISK-28822: chan_unistim.c: Recv error 9 (Bad file descriptor)
ASTERISK-28823: Updates for outgoing registrations not sent to the correct network address
ASTERISK-28824: BuildSystem: Search for Python/C API when possibly needed only.
ASTERISK-28825: Any curl response checks out as valid even if 404 is returned.
ASTERISK-28826: res_rtp_asterisk: Duplicate seqnos being added to send buffer with NACK
ASTERISK-28827: res_rtp_asterisk: Loop when receive buffer is flushed by a received packet that is also in receive buffer with NACK
ASTERISK-28828: NOTIFY sequence upsets MS Teams SIP trunk
ASTERISK-28829: app_queue: leaking stasis subscription when Redirecting call
ASTERISK-28830: Incorrect UTF-8 handling when using function FILTER
ASTERISK-28831: Leaking stasis subscriptions can linger indefinitely and brick Asterisk
ASTERISK-28832: chan_mobile creates PCMA streams that make some VoIP clients crash or not render received audio
ASTERISK-28833: SIP Hang- All the SIP peers are going unreachable at same time.
ASTERISK-28834: Segfault in taskprocessor_push
ASTERISK-28835: IPv6 addresses in SDP incorrectly formatted
ASTERISK-28836: chan_oss: Video Console broken.
ASTERISK-28837: pjproject_bundled: Honor --without-pjproject.
ASTERISK-28838: AST_MODULE_INFO requires, MODULEINFO does not mention
ASTERISK-28839: Sporadic crashes with Segmentation fault
ASTERISK-28840: samples: Missing for the modules cdr_ and cel_sqlite3_custom.
ASTERISK-28841: app_confbridge: Add support for disabling text messaging for a user
ASTERISK-28842: Hello
ASTERISK-28843: res_rtp_asterisk: Duplicate detection of DTMF - Wideband codec / MS customisation
ASTERISK-28844: samples: Start and Reload are blathering.
ASTERISK-28845: segfault and then crash
ASTERISK-28846: stream: Enforce formats immutability
ASTERISK-28847: ARI channels cuts the endpoint string over 80 characters
ASTERISK-28848: app_fax: Compile.
ASTERISK-28849: asttest failed to compile on fedora
ASTERISK-28850: sipp: Non-compliant XML files do not work with 3.6.0
ASTERISK-28851: Add useragent to CLI > 'pjsip show contacts'
ASTERISK-28852: Unprotected access to nochecksums variable, causes build failures
ASTERISK-28853: Missing include on FreeBSD
ASTERISK-28854: SIGSEGV when pjsip show history encounters IPV6 address
ASTERISK-28855: PJSIP - Implement CHANNEL(secure_bridge_media)
ASTERISK-28856: Codec Negotiation: Add incoming_call_answer_pref and outgoing_call_answer_pref
ASTERISK-28857: #exec call in pjsip.conf causes large delay (60 seconds) when reloading pjsip, partially locks up dialplan.
ASTERISK-28858: app_queue: Realtime linear queues losing the order of agents
ASTERISK-28859: pjsip: Increase maximum candidate count
ASTERISK-28860: pjsip: Resolve unsolicited->solicited aggregate issue
ASTERISK-28861: testsuite: Resolve unsolicited->solicited non-aggregate issue
ASTERISK-28862: res_musiconhold: Race condition between starting/stopping
ASTERISK-28863: The ast_rtp_codecs_payloads functions don't preserve order
ASTERISK-28864: RTP Timestamp not increasing after several transfers and codec changes
ASTERISK-28865: FAX T.38 re-Invite failed on '491 Another INVITE transaction in progress'
ASTERISK-28866: third-party/pjproject/configure.m4 contains bashisms
ASTERISK-28867: cannot get ANSWER Status from ${DIALSTATUS} though i get busy congested
ASTERISK-28868: app_alarmreceiver: does not call "eventcmd" with events as arguments or piping
ASTERISK-28869: pjsip: Crash in timer when sending request
ASTERISK-28870: streams: One memory leak and one issue cloning streams
ASTERISK-28871: res_pjsip_session: Unnecessary re-Invite on call answer
ASTERISK-28872: Asterisk services are not starting with ssl on 8089 port after configured http.conf
ASTERISK-28874: res_rtp_asterisk: RFC2833/RFC4733 Minimum signal duration not adhered to
ASTERISK-28875: All sip tranks are regularly unregister
ASTERISK-28876: Wrong next hop for INVITEs with PJSIP and PATH
ASTERISK-28877: Duplicate RINGING DeviceStateChange AMI events from chan_sip.c
ASTERISK-28878: chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16
ASTERISK-28879: pjproject has race conditions in it's build system
ASTERISK-28880: res_xmpp: Does not support urn:xmpp:ping causing session termination
ASTERISK-28881: res_pjsip_pubsub: Option to reduce log verbosity by selectively disabling missing/invalid subscriptions
ASTERISK-28882: res_pjsip: Contact not completely removed on transport closure
ASTERISK-28883: Spyee information ist missing in ChanSpyStop AMI Event
ASTERISK-28884: x-ast-orig-host not filtered out from request URI and To header
ASTERISK-28885: res_rtp_asterisk: Simultaneous termination and ICE complete can cause crash
ASTERISK-28886: chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
ASTERISK-28887: sendFAX fail with T.38
ASTERISK-28888: res_corosync: causes asterisk crash in huge distributed environment.
ASTERISK-28889: Outbound calls drops
ASTERISK-28890: res_pjsip_sdp_rtp: Keepalive not supported for video streams
ASTERISK-28891: documentation: AGICommand_set+music documentation arguments displayed incorreclty
ASTERISK-28892: res_musiconhold: Module res_musiconhold throws false warning
ASTERISK-28893: pbx_realtime: Cascading deadlock due to ast_autoservice_stop blocking
ASTERISK-28894: new memory leak when updating 16.9.0 to 16.10.0
ASTERISK-28895: res_pjsip_logger: Add tons'o'functionality
ASTERISK-28896: ari: Add support for specifying variables on channel create
ASTERISK-28897: app_confbridge: AMI Event "ConfbridgeTalking off" not fired when user leaves ConfBridge while talking
ASTERISK-28898: bridge_softmix: Conference bridge not passing silent rtp packets
ASTERISK-28899: Upgrade Asterisk to bundled pjproject 2.10
ASTERISK-28900: res_fax: Double frame free when gateway in use with off-nominal format usage
ASTERISK-28901: pjsip behaves incorrectly when sending RTP, it sends it to a private IP
ASTERISK-28902: High Memory consumption
ASTERISK-28903: res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
ASTERISK-28904: RTP ICE leaks the memory
ASTERISK-28905: Wrong Download Link On https://www.asterisk.org/download-asterisk-thank-you
ASTERISK-28906: ari: Race condition when destroying holding bridge multiple times with channels in it
ASTERISK-28907: Crash when removing stasis topic
ASTERISK-28908: pjsip_message_filter: 400 'Missing Contact header' reply to wrong port
ASTERISK-28909: chan_pjsip: CLI 'pjsip show channelstats' shows fax T.38 session as 'not valid' and doesn't print video stream stats
ASTERISK-28910: PJSIP: invalid value error exception when parsing 'Contact' header
ASTERISK-28911: Segmentation Fault on Voicemail Menu
ASTERISK-28912: Bug with a 'core show channels'
ASTERISK-28913: How to have digits spelled in different language
ASTERISK-28914: res_http_websocket: Client doesn't use mask
ASTERISK-28915: phone number porting
ASTERISK-28916: Memory leak with Asterisk 16 and malformed REGISTER requests
ASTERISK-28917: sip peers become unreachable
ASTERISK-28918: No Application SIPAddHeader()
ASTERISK-28919: we have been able to setup asterisk webrtc video call on both the endpoints but there is no audio from caller end but audio at receiver's end is working fine
ASTERISK-28920: bridge show all causes crash
ASTERISK-28921: Wrong return value check for fwrite when writing to pcap file
ASTERISK-28922: Attended transfer not swapping channel
ASTERISK-28923: T.38 Segfaults in chan_pjsip_queryoption
ASTERISK-28924: Imposible to add or read sip headers from a 302 Redirect
ASTERISK-28925: memory leak in asterisk 16.10
ASTERISK-28926: core dump trying to free null channel snapshot
ASTERISK-28927: Asterisk crash in music on hold
ASTERISK-28928: message proxy changes when client login with different IP
ASTERISK-28929: pjproject_bundled: Honor --without-pjproject.
ASTERISK-28930: ./configure --without-ssl build failure
ASTERISK-28931: stasis.c:1475 publish_msg: FRACK!, Failed assertion bad magic number 0x0 for object
ASTERISK-28932: res_pjsip_logger writing too big packets
ASTERISK-28933: res_pjsip.so fails to load when bundled pjproject is compiled without libssl
ASTERISK-28934: chan_mobile won't load without chan_alsa
ASTERISK-28935: app_meetme tries to create audio files with format sln
ASTERISK-28936: res_pjsip: crash when dialing non-sip uri
ASTERISK-28937: Task processor queue reached 500 scheduled tasks
ASTERISK-28938: core_unreal / core_local: Add support for multistream and re-negotiation
ASTERISK-28939: res_rtp_asterisk: Don't have send/receive buffers on non-WebRTC
ASTERISK-28940: /channels/create doesn't get any parameters from the body
ASTERISK-28941: segfault in pjsip timer
ASTERISK-28942: res_sorcery_memory_cache: Individual object expiration behaves unexpectedly with full backend caching
ASTERISK-28943: Asterisk can't start with the errors media_cache_item_populate_from_astdb: Unable to obtain information for file /tmp/...
ASTERISK-28944: bridge_softmix: Transitioning a stream from inactive -> sendrecv/sendonly doesn't re-negotiation
ASTERISK-28945: AMI SendText - add Content-Type parameter
ASTERISK-28946: Missing/wrong endpoint default values in configInfo documentation
ASTERISK-28947: Segmentation fault in mixmonitor_ds_destroy
ASTERISK-28948: ARI channel create doesn't referencing the channel_id parameter
ASTERISK-28949: res_http_websocket: Add masking to websocket client
ASTERISK-28950: Stale code in app_queue to check untouched channel
ASTERISK-28951: Inconsistent behaviour queues.conf when there is (not) a [general] section
ASTERISK-28952: Queue wrapuptime sometimes not respected (based on stale lastcall time)
ASTERISK-28953: res_pjsip_session: Preserve stream label
ASTERISK-28954: StreamEcho() only returns 1 active stream
ASTERISK-28955: "setvar" doesn't work properly in dahdi-channels.conf
ASTERISK-28956: res_odbc: ODBC connection does not always reconnect
ASTERISK-28957: chan_sip: chan_sip does not process 400 response to an INVITE.
ASTERISK-28958: Continue reading string when ping received by websocket
ASTERISK-28959: res_pjsip: Added option for disable rport parameter set
ASTERISK-28960: bridge: System gets into state where bridge is terminated after joining
ASTERISK-28961: res_pjsip_outbound_registration: Re-registration is incorrectly timed when response contains two identical Contact headers
ASTERISK-28962: Asterisk Memory Leak after SIP Reply Flood
ASTERISK-28963: Asterisk is killed when I connected to AMI
ASTERISK-28964: How to increase the ringback time of SIP endpoints registered to Asterisk
ASTERISK-28965: res_pjsip: Apply outbound proxy to static contacts on AOR
ASTERISK-28966: chan_pjsip: Interaction with provider re-INVITE and RTP causes codec flip-flop
ASTERISK-28967: Increase Agent Timeout in Queue
ASTERISK-28968: res_pjsip: Crash when comparing header in outgoing SIP request
ASTERISK-28969: res_pjsip: AMI command for show endpoint does not reflect active channels
ASTERISK-28970: Reflected XSS
ASTERISK-28971: app_userevent: Does not handle non-ASCII characters
ASTERISK-28972: FRACK! + task processor queue issue
ASTERISK-28973: Malformed IP address in SDP of 2nd SIP timer triggered INVITE when NAT is active (UDP transport with external_media_address)
ASTERISK-28974: res_rtp_asterisk: T.140 messages have appended RTP string to each message block.
ASTERISK-28975: res_http_websocket: Text payload data doesn't necessary include trailing zero
ASTERISK-28976: Crashes releted to pjsip
ASTERISK-28977: PJSIP can't SET CallerId
ASTERISK-28978: acl: named_acl rule misconfiguration results in segfault on reading rule from realtime
ASTERISK-28979: Not able to make call using PHP Agi
ASTERISK-28980: PJSIP outbound registration issue
ASTERISK-28981: I meet errors as I follow the pjsip setup scenario, how to fix it?
ASTERISK-28982: res_pjsip_t38: Does not resume as audio when negotiation fails
ASTERISK-28983: Unable to redirect outgoing calls to mobile
ASTERISK-28984: Asterisk suddenly crashs
ASTERISK-28985: ari: Sends channel event to multiple applications when connecting multiple over same socket
ASTERISK-28986: video over audio is not switching in webrtc with asterisk 16
ASTERISK-28987: BridgeCreated ARI event shows wrong video_mode info
ASTERISK-28988: Asterisk is crashing when we receive incoming calls
ASTERISK-28989: Asterisk is crashing when we receive incoming calls
ASTERISK-28990: chan_pjsip: Device state does not reflect hold
ASTERISK-28991: bridging: No channel is present when writing action
ASTERISK-28992: app_voicemail: Deadlock in ODBC when retrieving file
ASTERISK-28993: res_pjsip: Wrong Via and Contact is chosen, despite explicit configured transport
ASTERISK-28994: Limit Prefix
ASTERISK-28995: res_pjsip_registrar: Expires on statically configured contacts is not correct
ASTERISK-28996: chan_sip: TLS - Bad file descriptor errors
ASTERISK-28997: res_pjsip: Asterisk locks up and stops processing any SIP requests
ASTERISK-28998: Segfault in pj/timer.c
ASTERISK-28999: pjsip / sorcery / ao2: Large refcounts (FRACKs) on reload of large pjsip.conf configuration